Please help me choose the components for my dodecasub - Page 2 - AVS Forum
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post #31 of 41 Old 08-15-2012, 03:56 PM
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Originally Posted by cptomes 
I can put an arbitrary delay on any of the outputs of the dcx.

Interesting concept. I think I'm just going to have to try it.
A digital delay is fixed with respect to frequency, which is generally what you want when aligning speakers since they don't tend to change position with frequency. But wavelength varies with frequency, thus fixed delay means varying phase shift with frequency. That won't achieve the desired goal here. You would create the equivalent of two monopoles spaced by some distance, which might smooth room response or not for a given location, but would also have some wonky interactions happening inside the enclosure.

An all pass filter, usually implemented in analog form, is a "poor man's" version of the delay when digital processing isn't available. This phase shift is frequency dependent, but can work OK for sub alignment since you can align phase at the crossover point, above and below which the crossovers filter one speaker or another reducing the effect of destructive interference away from the xo point.

But this "good enough" filter is just what we need here, because it has another useful property in that the filter shelves at 180 and 0 degrees, just like a low or high pass frequency shelving filter is flat above and below certain frequencies. That allows the sub to be 180 out of phase above a certain frequency and stay there... and be 0 degrees out of phase below a certain frequency and stay there. Dipole above some frequency, monopole below some lower frequency, with a smooth transition between.

I don't remember if the behringer allows all pass filter topologies. Generally not needed when you have digital delay available. A quick search or look through the user manual should answer that question.


To LTD, what you are after is something like a shelving filter with fixed offset shelf. I'm not sure if that is possible in analog implementation, but I doubt it without getting into multitap stuff. I'm not an EE, always have to look this stuff up. Probably possible with digital filters, I suspect requiring FIR as it is akin to linear phase crossover topology.

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post #32 of 41 Old 08-15-2012, 03:59 PM
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Cptomes, see my above correction to the post you quoted. I spoke before turning on the aching brain.

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post #33 of 41 Old 08-16-2012, 05:19 PM
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Quote:
Originally Posted by Bigus View Post

A digital delay is fixed with respect to frequency, which is generally what you want when aligning speakers since they don't tend to change position with frequency. But wavelength varies with frequency, thus fixed delay means varying phase shift with frequency. That won't achieve the desired goal here. You would create the equivalent of two monopoles spaced by some distance, which might smooth room response or not for a given location, but would also have some wonky interactions happening inside the enclosure.
An all pass filter, usually implemented in analog form, is a "poor man's" version of the delay when digital processing isn't available. This phase shift is frequency dependent, but can work OK for sub alignment since you can align phase at the crossover point, above and below which the crossovers filter one speaker or another reducing the effect of destructive interference away from the xo point.
But this "good enough" filter is just what we need here, because it has another useful property in that the filter shelves at 180 and 0 degrees, just like a low or high pass frequency shelving filter is flat above and below certain frequencies. That allows the sub to be 180 out of phase above a certain frequency and stay there... and be 0 degrees out of phase below a certain frequency and stay there. Dipole above some frequency, monopole below some lower frequency, with a smooth transition between.
I don't remember if the behringer allows all pass filter topologies. Generally not needed when you have digital delay available. A quick search or look through the user manual should answer that question.
To LTD, what you are after is something like a shelving filter with fixed offset shelf. I'm not sure if that is possible in analog implementation, but I doubt it without getting into multitap stuff. I'm not an EE, always have to look this stuff up. Probably possible with digital filters, I suspect requiring FIR as it is akin to linear phase crossover topology.

I need to read up on how the DCX implements its filters again, but iirc they use digital processing to implement traditional analog filters (with frequency dependent phase shift, etc).

The test articles I would use for different signals (other than all same or some inverted) would have separate chambers for each driver and an amp channel on each driver.
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post #34 of 41 Old 08-18-2012, 02:11 AM
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Originally Posted by cptomes View Post

I need to read up on how the DCX implements its filters again, but iirc they use digital processing to implement traditional analog filters (with frequency dependent phase shift, etc).
DCX filters are implemented exactly the same as the equivalent linear circuit done in the analogue domain.
Cheapest stand alone FIR based unit would be the DEQX, but having heard the difference between the DEQX with its own extremely steep filters and the same DEQX using LR8 on the same 5 way system (uses 2 DEQX), the difference was audible, but very subtle with complex full range music. Difference with a single sub to mains xover at LF would be irrelevant in terms of audible difference.
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post #35 of 41 Old 08-18-2012, 09:10 AM
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Originally Posted by A9X-308 View Post

DCX filters are implemented exactly the same as the equivalent linear circuit done in the analogue domain.
Cheapest stand alone FIR based unit would be the DEQX, but having heard the difference between the DEQX with its own extremely steep filters and the same DEQX using LR8 on the same 5 way system (uses 2 DEQX), the difference was audible, but very subtle with complex full range music. Difference with a single sub to mains xover at LF would be irrelevant in terms of audible difference.

That's what I remembered. I stopped researching when I knew for sure that the filters behaved just like what I was used to using in analog setups. I used dbx crossovers before I got the DCX (still have 2 of them).
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post #36 of 41 Old 08-18-2012, 09:18 AM
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Originally Posted by A9X-308 View Post

DCX filters are implemented exactly the same as the equivalent linear circuit done in the analogue domain.
Cheapest stand alone FIR based unit would be the DEQX, but having heard the difference between the DEQX with its own extremely steep filters and the same DEQX using LR8 on the same 5 way system (uses 2 DEQX), the difference was audible, but very subtle with complex full range music. Difference with a single sub to mains xover at LF would be irrelevant in terms of audible difference.

Those are some steep filters. I played with basically every crossover topology available in the DCX and ended up with three I use on the same amps/drivers, depending on the listening I'm going to do. For low level (70dB spl at the listening position max) I've got a nice 6dB/octave setup. For moderately loud (~85dB) I use a 12dB/octave LR setup. For times when I need to pound something out of my brain, I switch to 24dB/octave LR. That setup is still not power-limited, the amps aren't clipping and we're still >3dB within the power handing of the drivers. I'm limited on the bottom end by excursion on the powered bass I use. Still gets me to ~100dB at the listening position...

How does the DEQX compare to the DCX? Is the DCX a subset of the functions that the DEQX has? Might be time to get a new set. Though I still haven't actually put my second DCX into use. No time to play...
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post #37 of 41 Old 08-19-2012, 12:12 PM
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Quote:
Originally Posted by cptomes View Post

That's what I remembered. I stopped researching when I knew for sure that the filters behaved just like what I was used to using in analog setups. I used dbx crossovers before I got the DCX (still have 2 of them).
The transfer function of the filter stages is equivalent, but the DCX offers delay and a ton of EQ capability as well, compared to an analogue offering the same slopes. The difference between a DCX and a Driverack is minimal.

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Those are some steep filters.
Not really - the DEQX can do 300dB/oct.
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I played with basically every crossover topology available in the DCX and ended up with three I use on the same amps/drivers, depending on the listening I'm going to do. For low level (70dB spl at the listening position max) I've got a nice 6dB/octave setup. For moderately loud (~85dB) I use a 12dB/octave LR setup. For times when I need to pound something out of my brain, I switch to 24dB/octave LR. That setup is still not power-limited, the amps aren't clipping and we're still >3dB within the power handing of the drivers. I'm limited on the bottom end by excursion on the powered bass I use. Still gets me to ~100dB at the listening position...
This makes no sense at all to me - I want the above band breakup of a sub driver to be seriously attenuated, no matter the listening level.
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Originally Posted by cptomes View Post

How does the DEQX compare to the DCX? Is the DCX a subset of the functions that the DEQX has? Might be time to get a new set. Though I still haven't actually put my second DCX into use. No time to play...
The IIR filters in a DCX are easily implemented compared to the FIR in a DEQX, so I guess the designer added them in to allow comparison between them easily. The DEQX can do everything the DCX can, plus more and is far easier to adjust through the software interface it uses.
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post #38 of 41 Old 08-19-2012, 06:48 PM
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Originally Posted by A9X-308 View Post

..
This makes no sense at all to me - I want the above band breakup of a sub driver to be seriously attenuated, no matter the listening level...

The lowest frequency driver in the system isn't a sub, it's running 30-100Hz. At higher levels I want to keep from hurting the drivers with out of band LOW frequency signals.

Different filters and alignments sound different. Even when adjusted to be pretty close to flat response, they still sound different. LR crossovers ring. Digitally implemented LR crossovers ring too.

I think I'll read up on the DEQX and see if it's time to upgrade.
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post #39 of 41 Old 08-19-2012, 07:29 PM
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Originally Posted by cptomes View Post

The lowest frequency driver in the system isn't a sub, it's running 30-100Hz. At higher levels I want to keep from hurting the drivers with out of band LOW frequency signals.
It's still low midbass at best, and any breakup not attenuated by a low order filter will still be audible in the passband of the driver above it as will distortion from the lower (LMB) driver. Signals from the LMB driver will cause some degree of combing when combining with the higher driver as frequencies increase, depending upon C-C distance. The >00Hz driver will also distort more in passband as it will still be reproducing lower bass signals as a not much reduced level.
I've been working with active filters for a long time now.
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LR crossovers ring. Digitally implemented LR crossovers ring too.
You're going to need to provide a credible reference for that one as I don't see how a Q=0.5 filter rings.
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post #40 of 41 Old 08-20-2012, 10:05 AM
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Transient response group delay is doubled for LR8 wrt LR4. Since we're talking about resonating ("ringing") circuits used as filters, I think "ring" isn't a completely inappropriate term. It's probably not one typically used when talking about this, but a spade is a spade and ignoring the behavior of a circuit just because of semantics is kind of silly.

http://www.rane.com/note160.html (found this after about 15 seconds of Googling btw, I'm sure there are others)

Towards the bottom of the page.

Since music has lots of transients, I disagree that it is inaudible.

The very best sounding speakers I have ever heard were Dunlavy SC-IVA. Close second was the SC-III. Both used first order crossovers with very close attention paid to transient response and group delay. Too bad the new owner closed down the company when he realized it wasn't going to make enough money for his tastes.
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post #41 of 41 Old 03-07-2013, 06:12 AM
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Since breakup is nonlinear, and depends on both frequency AND amplitude, my comment about different crossovers for different spl listening sessions makes sense. Why do you think PA systems use the highest crossover slopes available, yet home/enthusiast systems typically don't? Because for PA systems, running for hours at close to the drivers long-term power handling ability is not at all unusual. You need that steep slope to keep as much of the out of band signal away from drivers that will put out huge distortion and even self-destruct when driven with a fraction of the power they are rated to handle in-band.

With drivers of just average sensitivity, if you run them close to their power handling limit in a typical room in a typical private residence, you're going to have enough SPL to damage hearing or at least piss off the neighbors if you're in an apartment or zero lot line property. Typical long term power handling, ime, is NOT what results in low distortion, pleasing sound, but what the drive can handle DESPITE making huge amounts of nasty noise.

Different spl also puts you in different parts of the 3-d (not 2-d) Fletcher-Munson curves, so adjusting an active crossover system for flat response at a particular spl will result in a different sounding system when listened to at a significantly different spl (mostly from midbass and lower frequency) unless you trust a dynamic eq to not behave badly and know how to set it up. Since we're now talking about the nonlinearities in human hearing and perception, a "flat" measuring system is not the issue.
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